search for: ast_answ

...= ast_read_threadstorage_callid(); ^ chan_bridge_media.c:135:33: error: use of undeclared identifier 'pvt' chan = ast_unreal_new_channels(pvt, tech, ^ chan_bridge_media.c:141:13: error: use of undeclared identifier 'pvt' ast_answer(pvt->owner); ^ chan_bridge_media.c:142:13: error: use of undeclared identifier 'pvt' ast_answer(pvt->chan); ^ chan_bridge_media.c:144:34: error: use of undeclared identifier 'pvt' if (ast_channel_add_bridge_role(pvt->c...

...lly sent when the incoming leg to the pbx answered. It does not get delivered to the originating span (the pbx). It sees neither the call proceeding, alerting or any other of the indications that should have been sent. This causes the originating channel not to be in state AST_STATE_RINGING in ast_answer() called from ast_bridge_call(). The answer does not get propagated down to the originating channel. **** further digging ***** The problem apperas to be that the ALERTING indication from the pbx on the inbound leg is not propagated back up to the inbound leg from the pstn. The error is on t...

...om/. These Asterisk releases improve compatibility with T.38 switchovers internally; fixes an issue with Asterisk using poll() on OSX systems when it should not; allows chan_h323 to be built against both OpenH323 and H323Plus libraries (while simplifying the build process); and improve behavior of ast_answer() which was problematic for T.38 re-INVITES and other sorts of channel operations. Additionally, other bugs have also been resolved in these release candidates. The Asterisk-Addons release candidates fix a few minor issues since the last set of release candidates. The first beta release of the...

...Also, this Asterisk release improves compatibility with T.38 switchovers internally; fixes an issue with Asterisk using poll() on OSX systems when it should not; allows chan_h323 to be built against both OpenH323 and H323Plus libraries (while simplifying the build process); and improve behavior of ast_answer() which was problematic for T.38 re-INVITES and other sorts of channel operations. Additionally, other bugs have been resolved in this release. For a summary of the changes in this release, please see the release summary: http://svn.digium.com/svn/asterisk/tags/1.6.0.7/asterisk-1.6.0.7-summary....

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

...if (conf) { ao2_lock(conf); count = conf->markedusers; ao2_unlock(conf); }else{ count = 0; } if (!ast_strlen_zero(args.varname)) { snprintf(val, sizeof(val), "%d", count); pbx_builtin_setvar_helper(chan, args.varname, val); } else { if ( ast_channel_state(chan)!= AST_STATE_UP) { ast_answer(chan); } res = ast_say_number(chan, count, "",ast_channel_language(chan),(char *) NULL); } return res; } static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); res |= ast_register_application_xml(app2,count_exec); return res; } static int unload_module(void...

I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

...if (conf) { ao2_lock(conf); count = conf->markedusers; ao2_unlock(conf); }else{ count = 0; } if (!ast_strlen_zero(args.varname)) { snprintf(val, sizeof(val), "%d", count); pbx_builtin_setvar_helper(chan, args.varname, val); } else { if ( ast_channel_state(chan)!= AST_STATE_UP) { ast_answer(chan); } res = ast_say_number(chan, count, "",ast_channel_language(chan),(char *) NULL); } return res; } static int load_module(void) { ast_verb(3 ,"==Inside load_module=="); res |= ast_register_application_xml(app2,count_exec); return res; } static int unload_module(void...

Hello-- Why do I target chan_sip for so much effort? Because, it seems to me, chan_sip is probably the most used channel driver in the asterisk community!! (and, of course, the zap/dahdi driver, is also pretty popular) I haven't had time to follow up on chan_sip, and I probably won't for several months. But, if I had time, here is what I'd do: There are two ways to speed up